Acoustic system for detecting and locating low intensity and low frequency sound sources and related locating method

ABSTRACT

An acoustic system and method detect and locate low intensity and low frequency sound sources in an investigation area. An acoustic system is effective in identifying survivors trapped under rubble following a catastrophic event. The acoustic system focuses on the low frequency components of the human voice and includes acoustic sensors for detecting acoustic signals generated by the sound sources and for generating data representative of the acoustic signals. A wireless transmitter transmits the data representative of the detected acoustic signals to an electronic receiver block that receives and analyzes the data. A processor executes calibration of the acoustic sensors of the suite to temporally align each signal received from the acoustic sensors, and executes a digital beamforming to combine the data representative of the detected acoustic signals and to create an acoustic image of the investigation area to locate the low intensity and low frequency sound sources.

BACKGROUND ART OF THE INVENTION Field of Application

The present invention relates to an acoustic system for detecting andlocating low intensity and low frequency sound sources and to a relatedlocating method.

The invention in particular meets the need to make available an acousticsystem which is effective in identifying survivors trapped under therubble of buildings following the occurrence of a disastrous event, e.g.an earthquake, in which the presence and the position of such trappedpeople may be detected by the requests for help thereof.

Prior Art

Many companies today propose commercial acoustic arrays or microphonearrays which can be used in the field of passive audio imaging. Amongthese, the following are worth noting: Norsonic, CAE Systems, AcousticCamera, Dual Cam.

Such acoustic arrays allow the listening direction to be modifiedwithout physically moving the array itself by combining the signalsreceived from the individual microphones for each direction involved. Inparticular, a sound power for generating an acoustic image which isplaced over the optical image may be measured for each direction bymeans of scanning an optical image made available by an optionalcamcorder along the main directions of such an image.

An assembly comprising a microphone array and optionally a camera ordigital camcorder is commonly called an “acoustic camera”.

Commercially-used acoustic cameras of known type are characterized by amaximum aperture, i.e. by a maximum longitudinal dimension of about twometers. Such acoustic cameras are configured to sufficiently accuratelylocate a sound source having frequencies greater than 2 kHz.

There is a need to use acoustic cameras with broader aperture in orderto accurately locate sound sources at lower frequencies, and inparticular human voices. In particular, there is a need to use anacoustic camera with aperture of at least ten meters to detect a soundsource having frequencies in the order of 300 Hz.

An acoustic network system called ACOUFIND can be used for detectingtrapped people. Such an acoustic system is described in document:Menachem Friedman, Yoram Haddad, Alex Blekhman, “ACOUFIND: AcousticAd-Hoc Network System for Trapped Person Detection”, IEEE InternationalConference on Microwaves, Communications, Antennas and ElectronicSystems (COMCAS 2015), 2-4 Nov. 2015, Tel Aviv, Israel.

The ACOUFIND acoustic system consists of three main components:

a plurality of smartphones used as microphone sensors to record acousticdata in a specific known place;

a computer which collects acoustic data from all the smartphonerecordings and then analyzes them by processing the digital signal tocalculate the position of the survivor; and

a router which makes the connection between the smartphones and thecentral computer.

Such a system does not use a consistent acoustic array and theassessment of the originating direction of the sound source is obtainedwith triangulation techniques.

The need is particularly felt today to realize the acoustic detectionand locating of weak sound sources, i.e. low intensity and amplitude andlow frequency sound sources, in a preset area, in particular followingthe occurrence of a disastrous event in such an area, such as e.g. anearthquake.

As mentioned above, the solutions which can be used today for thispurpose are represented by the acoustic cameras and by the ACOUFINDacoustic system.

However such known solutions have limits and drawbacks.

Indeed, the commercial acoustic cameras are inadequate for theabove-indicated application due to the reduced aperture thereof (maximumof two meters). Such a reduced aperture causes a low angular resolutionwhich makes such acoustic cameras incapable of locating, in asufficiently accurate manner, the originating direction of low frequencyacoustic signals, i.e. of acoustic signals having frequencies less than1000 Hz.

The ACOUFIND acoustic system instead has the drawback that the abilityto detect the sound source is limited by the sensitivity of themicrophone installed on the smartphones.

SUMMARY OF THE INVENTION

It is the object of the present invention to devise and make available adeployable and reconfigurable lightweight acoustic system and a relatedmethod for detecting and locating low intensity and low frequency soundsources, which allows overcoming the above limitations in reference tothe known solutions.

The invention makes available an acoustic system which preferably, butnot exhaustively, can be used to find survivors under the rubble, forexample of a building, following the occurrence of a disastrous event,e.g. an earthquake, in which the presence and the position of thetrapped people may be detected on the basis of the requests for helpthereof.

In greater detail, it is the object of the invention to make availablean acoustic system for detecting and locating low intensity and lowfrequency sound sources having increased sensitivity to weak signalsthan what can be obtained with the single-microphone solution ofACOUFIND and furthermore, which can be configured to ensure an increaseddirectional accuracy with respect to an acoustic camera of commercialtype.

The invention proposed focuses on the low frequency components of thehuman voice, i.e. the components of the voice having frequency less than1000 Hz. Such components of the voice indeed show increased energycontent and are more resistant to the attenuation of the transmissionmeans.

The acoustic system of the invention in particular implements a broaderacoustic array than the known solutions, i.e. having aperture greaterthan 3 meters, preferably in the 10 to 20 meter range for optimalresults, for specifically detecting the spectral components of the humanvoice with frequency less than 1000 Hz.

Moreover, the invention achieves a virtual acoustic camera because it isconfigured to temporally align the microphones, thus increasing thesensitivity.

The acoustic system of the invention results in an increased detectionability and locating accuracy with respect to the more modern sensorsand acoustic systems while achieving an equivalent acoustic array havingwidth greater than 3 meters, preferably width in the range of 10 to 20meters for optimal results, and therefore by applying classical digitalbeamforming techniques.

Such beamforming techniques use algorithms for processing the acousticsignal which allow a receiver soundbeam to be formed by spatiallyisolating a sound originating from a given direction.

The specific application in environmental contexts such as thosegenerated by the occurrence of a disastrous event, e.g. an earthquake,require the use of a lightweight solution which can easily be deployedand configured with respect to the environment. The system of theinvention in particular does not need the sensors or microphones of theacoustic system to be arranged on a flat surface, rather allows a quickdeployment of the microphones themselves, which is independent of theposition thereof and does not affect the performance of the system. Forthis purpose, the acoustic system of the invention comprises an assemblyof acoustic sensors or microphones configured to be located scattered inan area, which can be wirelessly connected to one another and arrangedaccording to simple rules.

To cause such an assembly of sensors to become a true acoustic array,the invention advantageously provides the application of a calibrationprocedure adapted to estimate the related position of the sensorsthemselves with respect to a reference, and to save these positions ascorrections to be applied in the listening step of the sound signals, inparticular low intensity and low frequency sound signals.

The object of the invention is achieved by means of a reconfigurableacoustic system for detecting and locating low intensity and lowfrequency sound sources according to claim 1.

Preferred embodiments of such an acoustic system are described in thedependent claims.

The object of the present invention is also a method for detecting andlocating low intensity and low frequency sound sources using acalibration process according to claim 10.

BRIEF DESCRIPTION OF THE DRAWINGS

Further features and advantages of the acoustic system and of the methodaccording to the invention will become apparent from the followingdescription of preferred embodiments thereof, given only by way ofnon-limiting, indicative example, with reference to the accompanyingdrawings, in which:

FIG. 1 diagrammatically shows an acoustic system for detecting andlocating low intensity and low frequency sound sources according to theinvention;

FIG. 2 graphically shows an example of formation of a passive audioreceiver beam with respect to a rectangular scene;

FIGS. 3A to 3B graphically show steps of a known calibration methodwhich can be used in a linear acoustic array;

FIG. 4 shows, according to time, an example of test signal which can beused, in a calibration method, by the acoustic system for detecting andlocating low intensity and low frequency sound sources in FIG. 1;

FIG. 5 shows an autocorrelation signal generated from the test signal inFIG. 4, compared with a further autocorrelation signal obtained from arectangular signal;

FIG. 6 shows a first and second autocorrelation signal generated from asame test signal in FIG. 4, modulated in band frequency B=10 kHz andemitted by two separate sound sources;

FIG. 7 shows a simplified image of the acoustic system for detecting andlocating low intensity and low frequency sound sources in FIG. 1.

Equal or similar elements are indicated with the same numerals in theaforesaid FIGS. 1 to 7.

DETAILED DESCRIPTION

With reference to FIGS. 1 and 7, the acoustic system for detecting andlocating low intensity and low frequency sound sources in aninvestigation area IN according to the invention, is indicated as awhole with numeral 100.

Such an acoustic system 100 comprises a suite 1 of acoustic sensors orreceivers 2. Such a sensor suite 1 in particular comprises M groups 4 ofsensors 2 distributed in an investigation area IN, in which the sensorsuite 1 has a width or aperture AP.

Each sensor group 4 of the suite preferably, but not exhaustively,includes N acoustic sensors 2, for example N microphones verticallyarranged along a common support 6, for example a microphone supportconnected to ground 7. Thereby, the acoustic system 100 comprises asuite 1 including M×N acoustic sensors 2.

The acoustic sensors 2 of the suite in particular may be randomlydistributed in the investigation area IN to detect acoustic signalsgenerated by the low intensity and frequency sound sources and togenerate data representative of such acoustic signals. The acousticsensors 2 of the suite practically are arranged arbitrarily in theinvestigation area IN, i.e. they are not bound to belonging to aspecific surface or structure. The causal term related to thedistribution of the acoustic sensors 2 of the suite in the presentinvention means that the sensors are distributed in a “suitable” manner,i.e. without being subjected to a specific rule or obligation, but so asto follow the needs of the end user, without however geometric typelimits or obligations. This increases the flexibility of use of theacoustic system 100 of the invention.

Width or aperture of suite 1 of acoustic sensors 2 later in thisdescription means the distance as the crow flies between the two groups4 of sensors 2 the furthest from each other.

Moreover, the groups 4 of sensors 2 of suite 1 are not necessarilyaligned on ground 7 along the same direction, as shown in FIG. 1, butthey may be randomly distributed in the investigation area IN. In anycase, such sensor groups 4 are positioned within a sphere having adiameter which is equal to the aforesaid width or aperture AP of thesensor suite 1.

In one example embodiment, such a width or aperture AP of suite 1 ofsensors 2 is greater than 3 meters. In a preferred example embodiment,such a width or aperture AP of the sensor suite 1 is in the range of 10to 20 meters for optimal results.

In other words, the number of microphones 2 used being equal, system 100of the invention allows a plurality of various configurations to beachieved by simply deploying the microphones 2 according to differentlayouts of the suite, each of which has a respective aperture AP. On thebasis of simulations and experimental tests, the Applicant has seen thatsystem 100 ensures an increased sensitivity to various parts of theacoustic spectrum with the above-mentioned apertures AP of suite 1 ofsensors 2.

Moreover, the acoustic system 100 comprises an electronic calibrationapparatus 5, i.e. an apparatus operating as sound source and configuredto calibrate suite 1 of the acoustic sensors.

Indeed, in order to quickly distribute the acoustic sensors 2 of theacoustic system 100 in the investigation area IN, such acoustic sensors2 may be positioned according to an approximate “practical rule”. Inreference to the example in FIG. 1, it is provided for the distancebetween the two groups 4 of sensors 2 furthest from each other to beAP=10 and for the remaining groups 4 of sensors 2 to be more or lessuniformly positioned in intermediate position between such furthestgroups of sensors.

It is worth noting that the low intensity and low frequency sound sourcefor the objects of the present invention is the human voice havingfrequency components less than 1000 Hz.

The acoustic system 100 further comprises wireless communication means15, in particular a wireless link, to allow a wireless typecommunication between suite 1 of sensors 2 and the electroniccalibration apparatus 5 on the one hand, and an electronic receiverblock 20, in particular a multi-channel block, and a processing unit 30of the acoustic system 100 on the other hand.

In greater detail, through the aforesaid wireless link 15, the acousticsystem 100 is configured to:

-   -   enable the transmission of data acquired by suite 1 of sensors        2, to the multi-channel receiver block 20;    -   execute a control and time alignment of the sensors 2 of the        suite.

The above-mentioned multi-channel electronic receiver block 20 isconfigured to receive and analyze the data representative of thedetected acoustic signals. Such a multi-channel receiver block 20 inparticular comprises a wireless multilink and a processing block, bothmade using standard hardware components. The multi-channel receiverblock 20 further comprises software components configured to collectdata representative of acoustic signals acquired by suite 1 of acousticsensors 2 and to sample such signals.

The acoustic system 100 further comprises the above-mentioned processingunit 30 operatively associated with the multi-channel electronicreceiver block 20. Such a processing unit 30, which for examplematerializes in a personal computer (PC), is configured to execute insequence:

-   -   an operation of “calibrating and controlling” acoustic sensors 2        of suite 1 to temporally align the acoustic signals received        from the acoustic sensors 2 randomly distributed in said        investigation area IN,    -   an operation of “digital beamforming” to combine the data        representative of the acoustic signals detected by the acoustic        sensors 2 and to create an acoustic image of the investigation        area to locate said low intensity and low frequency sound        sources.

Indeed, to cause the above-mentioned assembly of sensors 2 to become atrue acoustic array, the aforesaid electronic calibration apparatus 5,which is operatively associated with the processing unit 30, isconfigured to emit suitable sound waveforms on the basis of indicationsand instructions imparted by the processing unit 30 itself.

The execution of an appropriate calibration procedure by the processingunit 30 allows the related position between each sensor 2 of array 1 tobe estimated in the three-dimensional space 3D (x, y, z).

The processing unit 30 in particular is adapted, through the“calibration and control” operations, to manage the calibrationprocedure and to process the data acquired by suite 1 of sensors 2 whenthey are stimulated by the calibration apparatus 5, as is clarifiedbelow.

The acoustic system 100 requires for the aforesaid calibration operationto be executed at least once following the deployment of suite 1 ofsensors 2 in the investigation area IN.

Following such a calibration operation, the acoustic system 100 isconfigured to use beamforming techniques, which are known to an expertin the field, on the sound signals detected by the acoustic array 1.

The processing unit 30 is adapted, through such “digital beamforming”operations, to implement the passive audio imaging by forming receiverbeams (as shown in FIG. 2) for all the directions involved through asingle acquisition of sound signals.

In greater detail, such a FIG. 2 diagrammatically shows the acousticarray 1, which distance from the scanning plane 50 is indicated by anarrow F, and a beam model 40 emitted by such an acoustic array 1.

An example embodiment of a calibration method executed by the acousticsystem 100 for detecting and locating low intensity and low frequencysound sources proposed is described below in reference to FIGS. 3A, 3B4, 5 and 6.

As is known, calibration is an operation which becomes necessary inorder to apply the classical beamforming algorithms, e.g. “dataindependent” algorithms, on an array of acoustic sensors or receivers 2randomly positioned in an investigation area, such as the acoustic array1 in FIG. 1.

In particular, when an assembly of sensors 2 is arranged to form anarray, there is a need to know the geometry of the array, i.e. therelated position of each sensor of the array with respect to the others.

In the case of acoustic camera, the microphones of the array arepositioned on a frame to form a rigid structure having substantiallyplanar shape. Therefore, the position of each sensor in the array isknown beforehand.

In the case instead of a layout of the microphones 2 over aninvestigation area IN, such microphones 2 in general are arrangedrandomly with respect to a reference plane according to the specificityof the environment observed and the type of information to be received.

In general terms, according to a known calibration method referring to abi-dimensional array, in particular a linear array like the one shown byway of example in FIG. 3A, it is assumed that one of the sensors orreceivers, e.g. sensor r4, is the reference sensor (or pivot sensor). Afirst sound source A is arranged at a known distance d from a straightline X which crosses such a reference sensor r4 and is orthogonal to thesegment having the same reference sensor r4 and the first source A asends.

The secondary sensors r1, r2, r3, r5, r6, 5 r7 of the array arespatially spaced apart from the straight line X.

As is known, the object of the calibration procedure is to measure thepositions of these secondary sensors r1, r2, r3, r5, r6, r7 in thereference system A-r4-X defined by the first source A, by the referencesensor r4 and by the straight line X, and to estimate the corrections tobe applied to each of such secondary sensors in terms of positive ornegative time delay to be applied to the detected sound signals tosimulate an aligned array.

Again with reference to FIG. 3A, a second sound source B is consideredfor this purpose in addition to the first sound source A and to thereference sensor r4. The position of one of the secondary sensors in thereference system A-r4-X, e.g. secondary sensor r5, is obtained bymeasuring a first dr5A and a second dr5B distance of such a secondarysensor r5 from the first A and from the second B sound source,respectively.

The procedure then provides calculating the equations of twocircumferences C1, C2 with middle in the position of the first A and ofthe second B sound source and having the first dr5A and the second dr5Bdistance, respectively, as radius. The position of the secondary sensorr5 is obtained by intersecting the equations of the aforesaidcircumferences C1 and C2.

Once the position of the secondary sensor r5 is known the calibrationprocedure provides a step of calculating a time shift Δt5 representativeof the spatial distance between the position of the secondary sensor r5and the projection r5 of the position of such a sensor orthogonal to thestraight line X in order to align all the elements of the array.

In reference to FIG. 3B, by repeating the aforesaid operations for allthe secondary sensors of the linear array r1, r2, r3, r5, r6, r7, afirst time shift vector (Δt1, Δt2, . . . , Δt7) and a second vector maybe calculated, in which each element is representative of the differencebetween the actual position of the aforesaid secondary sensors r1, r2,r3, r5, r6, r7 and the position of such sensors aligned (along astraight line X) with the reference sensor r4, i.e. vector Δx1, Δx2, . .. , Δx7. Each time shift of the first vector (Δt1, Δt2, . . . , Δt7) isrepresentative of the greater or lesser distance of the secondarysensors r1, r2, r3, r5, r6, r7 from the alignment condition.

The above-described process for linear array may be extended to anactual case, i.e. to an array distributed in a three-dimensional space,such as array 1 in FIG. 1. The process here provides the use of threesound calibration sources.

The position in space of each sensor 2 of the acoustic array 1 can beobtained by measuring the three distances between a preselectedreference acoustic sensor (pivot sensor) and each of these three soundcalibration sources. Moreover, the intersecting point of three spheresis calculated, each having middle at one of the aforesaid sound sourcesand the three distances measured as radius.

Looking in greater detail at the calibration method implemented in theacoustic system 100 of the present invention, the same acoustic wavesare used to measure the distances between the aforesaid three soundcalibration sources and the sensors or microphones 2 of suite 1. Oncethe speed of the sound in the air is known, a same waveform emitted by asound source is received at various distances at various times.Therefore, the measurement of the time shift is proportional to themeasurement of the distance of the various sensors 2 of array 1 from thereference sensor.

In order to increase the accuracy in measuring the distance, thecalibration method of the invention provides using test signals havingwaveforms modulated in “chirp” type frequency, i.e. signals in which thefrequency varies linearly over time by increasing or decreasing.

An example of test signal (S(t)) which can be used in the calibrationmethod of the invention is shown in FIG. 4.

Such a test signal (S(t)), in particular an increasing frequency signal,can be expressed by means of the equation:

$\begin{matrix}{{{s(t)} = {{\cos \left\lbrack {2{\pi \left( {{f_{0}t} + \frac{{Kt}^{2}}{2}} \right)}} \right\rbrack}\mspace{14mu} {with}}}{{{t} \leq {\frac{T}{2}\mspace{14mu} {and}\mspace{14mu} B}} = {KT}}} & (1)\end{matrix}$

where T is the duration of the modulated pulse and B is the frequencyrange of such a pulse. The use of the aforesaid frequency modulatingwaveforms (S(t)) increases the calibration accuracy.At least one of such test signals (S(t)) is emitted by each of the threesound calibration sources and is received by all the microphones 2 ofsystem 100. The three sound calibration sources in particular arelocated in the calibration apparatus 5.

During calibration, the test signal (S(t)) is emitted in sequence byeach of these three calibration sources with a preset transmission timeinterval with respect to the same signal emitted by one of the other twosources. Such a transmission time interval is for example, greater thanduration T of the modulated pulse.

The method provides a step in which the test signals (S(t)) emitted bythe three calibration sources and received by each microphone 2 of array1 are digitalized to generate respective digital signals Sd to be sentto the processing unit 30.

The method in such a processing unit 30 provides a step of executing aconvolution operation between each digital signal Sd received with acopy of the test signal (S(t)), which is saved in a respective memory ofthe processing unit 30.

Such a convolution operation generates an autocorrelation signal ACsimilar to the one shown in FIG. 5. Such a FIG. 5 in particular shows anexample of autocorrelation signal AC compared with a furtherautocorrelation signal AR obtained from a rectangular signal.

With reference to FIG. 5, it is worth noting that the width of the mainloop a −3 dB of the autocorrelation signal AC obtained from the testsignal (S(t)) is inversely proportional to the bandwidth B of such asignal.

Moreover, if the same test signal (S(t)) received from the sensors 2originates from a first and a second sound calibration source arrangedat a distance d1 from each other, the processing unit 30 is configuredto generate a first AC1 and a second AC2 autocorrelation signal having atime shift from each other which is directly proportional to distance d1between such sources. In greater detail, such a time shift is equal tothe ratio d1/vs, where vs is the speed of the sound.

By way of example, FIG. 6 shows the first AC1 and the second AC2autocorrelation signal generated from a same test signal (S(t)),modulated in band frequency B=10 kHz and emitted by two separate soundcalibration sources arranged at a distance from each other of d1=30 cm.In reference to the example in FIG. 6, the peak portions of theautocorrelation functions AC1 and AC2 are spaced apart from one anotherby about one millisecond. The width of the main loop of theautocorrelation at −3 dB for the aforesaid functions (obtained byintersecting the loop with a horizontal straight line passing through −3dB) is equal to about 0.1 milliseconds, corresponding to about 3 cm.

The peak position of the functions may therefore be estimated with anaccuracy equal to at least 3/√12, which is sufficient for theapplication proposed.Such a procedure, which is repeated for the signals emitted by the threecalibration sources and for each microphone 2 of array 1, allows theprocessing unit 30 to collect the estimated distances between the peaksof the related autocorrelation functions to generate a correctionmatrix.

The elements of the aforesaid correction matrix are representative ofthe correction to be applied to each microphone 2 and to the relatedreceiver channel prior to the beamforming step. In greater detail, suchcorrections are applied to the low intensity and low frequency soundsignals received and sampled by each receiver channel of the microphones2, and can be expressed as positive or negative time shift of samples ofsuch signals.

In other words, the calibration of the acoustic sensors 2 of suite 1comprises the following steps:

-   -   emitting in sequence, at preset time intervals, at least one        calibration test signal (S(t)) by the three sound calibration        sources of the electronic calibration apparatus 5;    -   receiving, by any one of the acoustic sensors 2 of suite 1, the        three test signals S(t) originating from each of said sound        sources to generate three digital signals Sd to be sent to the        processing unit 30;    -   executing, by the processing unit 30, a convolution between each        of the three digital signals Sd received with a copy of the test        signal S(t) emitted to generate three autocorrelation signals        AC;    -   calculating, from a known distance d1 between the sound        calibration sources, a respective distance between the portions        at −3 db of the three autocorrelation signals to estimate a time        shift between the samples of the received test signals S(t);    -   repeating said execution and calculation steps for each acoustic        sensor 2 of suite 1, and    -   generating, by the processing unit 30, a correction matrix which        elements are representative of corrections in the time shift        applicable to the samples of the signals emitted by low        intensity and low frequency sound sources received from each        acoustic sensor 2 of the suite 1.

The acoustic system 100 of the invention is configured, under operatingconditions, to implement and execute digital beamforming algorithms tostimulate a scanning of all the possible receiver directions of a soundsignal in order to detect and locate low intensity and low frequencysound sources, such as for example, voices or calls emitted by aindividuals trapped under the rubble of buildings following theoccurrence of a disastrous event, such as an earthquake.

As is known, the term beamforming is used to indicate a variety ofprocessing algorithms of the signals received by the microphones 2 of anacoustic array 1 so that the latter focuses on and faces a particulardirection without any intervention being carried out on the structure ofarray 1 or on the number or on the layout of the microphones 2 of thearray.

In particular, beamforming consists of a spatial filtering applicable toan array of elements, in the case under examination, microphones of anacoustic array, in light of the (spatial) diversity of the signalsreceived from each thereof.

According to the criteria with which the coefficients characterizing theaforesaid spatial filter are calculated, the beamforming algorithms maybe broken down into:

-   -   data-independent algorithms;    -   excellent algorithms from a statistical viewpoint.

The coefficients in data-independent algorithms do not depend on thedata and are selected so that the response of the beamformerapproximates a desired response known beforehand.

All the examples of beamforming algorithms described belowadvantageously can be implemented by the acoustic system 100 of theinvention.

Among the data-independent algorithms, one of the most common and soundsolutions is the Bartlett beamformer, or delay-and-sum (DAS)beamforming. In greater detail, the DAS beamforming algorithm isconfigured to apply a delay and an amplitude weight to the signalreceived from each element of the array, to then add all thecontributions thus processed. The delays are selected so as to maximumthe sensitivity of the array to the emissions originating from aparticular direction. Indeed, the suitable adjustment of the delaysallows the signals collected by the individual elements of the array tobe constructively added, in fact pointing the array in the direction ofa specific source. Further details on the DAS beamforming algorithm aredescribed in the document found at the link:

http://www.labbookpages.co.uk/audio/beamforming/delaySum.html

The coefficients of the spatial filter in the excellent algorithms froma statistical viewpoint are selected according to the statistics of thereceived data. The output provided by such algorithms also contains acontribution—albeit minimum—due to the interfering signals and to thenoise. These are adaptative algorithms, designed so the responseconverges with a statistically excellent solution.

The Minimum Variance Distortionless Response (MVDR) is worth notingamong the algorithms of this second category. Such an MVDR algorithm isadapted to minimize the variance of the outlet signal with linearpointing constraint. The signals received from the elements of the arrayin particular are used to calculate the spatial coefficients in order tomitigate the effect of the noise and of the interferences. Furtherdetails on the MVDR algorithm are described in the document found at thelink:

https://www.med.ira.inaf.it/Medichats/29_01_2008/Medichat_29.01.2008.ppt

In the specific case in which the geometry of the array is not fixed orknown, such as in the acoustic system 100 of the invention, thecalibration operations may be particularly burdensome from acomputational viewpoint due to a random distribution of the constitutingelements. “Blind beamforming” algorithms may be used in these cases,which exclusively use data collected by the elements of the array tofocus and direct the latter towards a particular direction. An exampleof beamforming algorithm comprising three randomly distributed sensorsis described in document YAO et al.: Blind Beamforming on a RandomlyDistributed Sensor Array System, IEEE JOURNAL ON 15 SELECTED AREAS INCOMMUNICATIONS, VOL. 16, NO. 8, OCTOBER 1998).

The advantages of the blind beamforming lie in the possibility ofignoring both the radiation diagram of the elements of the array and theinsensitivity to the calibration errors.

On the basis of what is disclosed above, it may be concluded that theacoustic system 100 for detecting and locating low intensity and lowfrequency sound sources of the invention and the related method areprovided with advanced functionalities for detecting the acousticsources and accuracy functionalities in locating persons, whilesimultaneously ensuring implementation speed, flexibility andsimplicity.

With respect to the known solutions, the acoustic system 100 of theinvention in particular includes the following innovative aspects:

-   -   digital beamforming techniques and algorithms in the “far field”        to combine the data representative of the acoustic signals        detected by the acoustic sensors 2;    -   creation of an acoustic image obtained by modifying the        listening direction of the acoustic array 1 and then        interpolating the power measured by each point of the scanning        plane 50; the acquired data set for each direction of the area        involved 50 in particular is processed to obtain the effect of a        direction microphone pointed in the generic direction;    -   use of acoustic sensors 2 having superior sensitivity features        with respect to the microphones currently used in other        applications, for example in smartphones.

Moreover, the acoustic system 100 of the invention advantageously:

-   -   realizes an acoustic array 1 having a broader aperture than the        one of the known solutions, in particular greater than 3 meters,        preferably in the range of 10-to 20 meters for excellent        results;    -   allows flexibility both in the dimensions and in the layout of        the microphone array, thus ensuring optimized performance with        respect to the environment observed;    -   ensures a quick layout of the acoustic sensors 2, given the        possibility of arranging the same substantially in random        manner.

The acoustic system 100 of the invention has the following advantages:

-   -   ensures improved locating sensitivity and accuracy with respect        to the devices currently available;    -   quick and simple and flexible implementation, which is a        significant requirement if trapped survivors are to be located        under the rubble of buildings following the occurrence of a        disastrous event, e.g. an earthquake;    -   a system which can be reconfigured in terms of dimensions and        layout of the elements, and which may also be positioned at a        safety distance from the area observed;    -   use of the acoustic technology as effective complement to radar        or seismic systems.

For example, radar detection systems suffer from noises originating fromsmall movements around the analyzed site (water flow, crowd movement,etc.) which may cause false alarms and problems of electromagneticinterference (including the cellular base stations).

Seismic systems suffer from noise caused by the vibrations of vehicles,machinery or persons and have limited accuracy.

Contrarily, the aforesaid limits are overcome with the presentinvention. The overall detection capacity of system 100 is improved bythe combination of the acquired signals originating from the acousticsensors 2 through digital beamforming techniques which increase thesensitivity of system 15 with respect to the individual microphones 2associated with array 1. In other words, the survivors who carry outcalls of weak intensity and/or furthest from array 1 may in any case bedetected with the acoustic system 100. Moreover, system 100 of theinvention is adapted to create a virtual acoustic camera by creating anacoustic image of a given site which may be placed over the opticalimage provided by an (optional) camcorder.

Those skilled in the art may make several changes and adaptations to theembodiments of the system and method of the invention, and may replaceelements with others which are functionally equivalent in order to meetcontingent needs, without departing from the scope of the followingclaims. Each of the features described as belonging to a possibleembodiment may be achieved regardless of the other embodimentsdescribed.

1. An acoustic system (100) for detecting and locating low intensity andlow frequency sound sources in an investigation area (IN), comprising: asuite (1) of acoustic sensors (2) randomly distributed in saidinvestigation area (IN) to detect acoustic signals generated by saidsound sources and to generate data representative of said acousticsignals; an electronic receiver block (20) configured to receive andanalyze the data representative of the detected acoustic signals;wireless communication means (15) for transmitting said datarepresentative of the acoustic signals detected by the acoustic sensors(2) to the electronic receiver block (20); a processing unit (30)operatively associated with the electronic receiver block (20), saidprocessing unit (30) being configured to execute in sequence: anoperation of calibrating the acoustic sensors (2) of the suite (1) totemporally align the acoustic signals received from the acoustic sensors(2) randomly distributed in said investigation area, an operation ofdigital beamforming to combine the data representative of the acousticsignals detected by the acoustic sensors (2) and to create an acousticimage of the investigation area to locate said low intensity and lowfrequency sound sources.
 2. An acoustic system (100) according to claim1, further comprising an electronic apparatus (5) for calibratingacoustic sensors (2), connected to said processing unit (30) through thewireless communication means (15) to receive start instructions of saidcalibration operation.
 3. An acoustic system (100) according to claim 1or 2, wherein said suite (1) of acoustic sensors (2) has a width (AP)which is greater than three meters.
 4. An acoustic system (100)according to claim 1 or 2, wherein said suite (1) of acoustic sensors(2) has a width (AP) comprised in the range of 10 meters to 20 meters.5. An acoustic system (100) according to claim 2, wherein saidelectronic calibration apparatus (5) is configured to emit, on the basisof the start instructions received from the processing unit (30), atleast one calibration test signal (S(t)) having a waveform with linearlyvariable frequency over time.
 6. An acoustic system (100) according toclaim 5, wherein said at least one calibration test signal (S(t)) has alinearly increasing frequency over time.
 7. An acoustic system (100)according to claim 5 or 6, wherein said electronic calibration apparatus(5) includes three sound calibration sources, each of which isconfigured to emit said at least one calibration test signal (S(t)). 8.An acoustic system (100) according to any one of the preceding claims,wherein said acoustic sensors (2) are microphones.
 9. An acoustic system(100) according to any one of the preceding claims, wherein the lowintensity and low frequency sound source is the human voice havingfrequency components less than 1000 Hz.
 10. A method for detecting andlocating low intensity and low frequency sound sources in aninvestigation area (IN), using an acoustic system (100) which comprises:a suite (1) of acoustic sensors (2), an electronic receiver block (20),wireless communication means (15), a processing unit (30) operativelyassociated with the electronic receiver block (20), the methodcomprising the steps of: randomly distributing the acoustic sensors (2)in said investigation area (IN); executing, by the processing unit (30),an operation of calibrating acoustic sensors (2) of the suite (1) totemporally align each acoustic signal received from the acoustic sensors(2); detecting, by the acoustic sensors (2), acoustic signals generatedby said sound sources and for generating data representative of saidacoustic signals; transmitting, by the wireless communication means(15), said data representative of the acoustic signals detected by theacoustic sensors (2), to the electronic receiver block (20); receivingand analyzing, by the electronic receiver block (20), the datarepresentative of the detected acoustic signals; executing, by theprocessing unit (30), a digital beamforming to combine the datarepresentative of the acoustic signals detected by the acoustic sensors(2) and to create an acoustic image of the investigation area to locatesaid low intensity and low frequency sound sources.
 11. A method fordetecting and locating low intensity and low frequency sound sources inan investigation area (IN) according to claim 10, wherein the acousticsystem further comprises an electronic apparatus (5) for calibratingacoustic sensors (2), connected to said processing unit (30) through thewireless communication means (15), said step of executing a calibrationof the acoustic sensors (2) of the suite (1) comprises a step ofemitting, by the electronic calibration apparatus (5), at least onecalibration test signal (S(t)) having a waveform with linearly variablefrequency over time.
 12. A method for detecting and locating lowintensity and low frequency sound sources in an investigation area (IN)according to claim 11, wherein said at least one calibration test signal(S(t)) has a linearly increasing frequency over time.
 13. A method fordetecting and locating low intensity and low frequency sound sources inan investigation area (IN) according to claim 11, wherein said step ofexecuting a calibration of the acoustic sensors (2) of the suite (1)further comprises the steps of: emitting in sequence, by three soundcalibration sources of the electronic calibration apparatus (5), said atleast one calibration test signal (S(t)) at preset time intervals;receiving, by any one of the acoustic sensors (2) of suite (1), thethree test signals (S(t)) originating from each of said sound sources togenerate three digital signals (Sd) to be sent to the processing unit(30); executing, by the processing unit (30), a convolution between eachof the three digital signals (Sd) received with a copy of the testsignal (S(t)) emitted to generate three autocorrelation signals (AC);calculating, from a known distance (d1) between the sound calibrationsources, a respective distance between the portions at −3 db of thethree autocorrelation signals to estimate a time shift between thesamples of the received test signals (S(t)); repeating said executionand calculation steps for each acoustic sensor (2) of the suite (1), andgenerating, by the processing unit (30), a correction matrix whichelements are representative of corrections in the time shift applicableto the samples of the signals emitted by low intensity and low frequencysound sources received from each acoustic sensor (2) of the suite (1).14. A method for detecting and locating low intensity and low frequencysound sources in an investigation area (IN) according to claim 10,wherein said step of randomly distributing the acoustic sensors (2)comprises a step of distributing said sensors so that the suite (1) hasa width (AP) which is greater than three meters.
 15. A method fordetecting and locating low intensity and low frequency sound sources inan investigation area (IN) according to claim 10, wherein said step ofrandomly distributing the acoustic sensors (2) comprises a step ofdistributing said sensors so that the suite (1) has a width (AP)comprised in the range of 10 meters to 20 meters.
 16. A method fordetecting and locating low intensity and low frequency sound sources inan investigation area (IN) according to claim 10, wherein the lowintensity and low frequency sound source is the human voice havingfrequency components less than 1000 Hz.